| Method and apparatus for dynamically adjusting the playout delay of audio signals -> Monitor Keywords |
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Method and apparatus for dynamically adjusting the playout delay of audio signalsRelated Patent Categories: Multiplex Communications, Pathfinding Or Routing, Combined Circuit Switching And Packet Switching, Routing Circuit Switched Traffic Through A Packet Switching NetworkMethod and apparatus for dynamically adjusting the playout delay of audio signals description/claimsThe Patent Description & Claims data below is from USPTO Patent Application 20070211704, Method and apparatus for dynamically adjusting the playout delay of audio signals. Brief Patent Description - Full Patent Description - Patent Application Claims
FIELD OF THE INVENTION [0001] The present invention generally relates to a real-time voice communication system, and more specifically to a method and apparatus for dynamically adjusting the playout delay of audio signals. BACKGROUND OF THE INVENTION [0002] As the Internet expands rapidly, the service of voice over IP (VoIP) is widely adopted. However, the network traffic conditions remain the most important factor for the voice quality of VoIP regardless of the compression techniques used. When the network latency varies, the packet containing the compressed voice data is delayed or even lost to reach the receiver end. For the VoIP application, the voice packet loss or out-of-order arrival will greatly affect the voice quality. [0003] In the VoIP system, the arrival time of the voice packets will be jittered due to the network delay variation. The current use of jitter buffer is the most widely employed technique for solving this problem. By storing the received voice packets in the jitter buffer to delay the playout, the network impact will be reduced on the playout voice quality. [0004] In the jitter buffer management mechanism, the delay length of the voice packets plays the key role in the voice quality. The current delayed playout designs are divided into two categories. The first is to use a fixed length (constant) delay in playout, and the second is to use an adjustable playout delay. FIG. 1 shows a schematic view of fixed playout delay. The small dots in the figure indicate the voice packets arriving at the receiving end. The x-axis is the arrival time in milliseconds (ms), and y-axis is the voice packet delay, that is, the transmission time of the voice packet in the network. The two horizontal lines in FIG. 1 are the 200 ms and 90 ms fixed playout delay, respectively. [0005] As shown in FIG. 1, the drawback of the fixed playout delay is that when the fixed playout delay is too small, such as 90 ms, some voice packets will arrive too late to be played back. This can be solved by a longer fixed playout delay. However, a longer fixed playout delay, such as 200 ms, will cause the degradation of the voice communication quality. [0006] The advantage of the fixed playout delay is the low computation complexity in the implementation, while the drawback is that it does not reflect the actual network conditions. Once the network is congested and the jitter buffer is overflow, the communication will be cut off. [0007] To solve the aforementioned drawback, related researches were conducted to develop adjustable playout delay techniques so that the delay can be adjustable in accordance with the network conditions by adjusting the jitter buffer size. A plurality of techniques are disclosed in related patents, including U.S. Pat. No. 6,360,271, U.S. Pat. No. 6,600,759, U.S. Pat. No. 6,693,921, U.S. Pat. No. 6,452,950, U.S. Pat. No. 6,700,895, U.S. Pat. No. 6,684,273, U.S. Pat. No. 6,683,889 and U.S. Pat. No. 6,747,999. [0008] U.S. Pat. No. 6,360,271 disclosed a "system for dynamic jitter buffer management based on synchronized clocks" to use a global positioning system (GPS) to synchronize the clock. By arranging the playout delay for each voice packet, the patent provides a dynamic jitter buffer management mechanism. [0009] U.S. Pat. No. 6,600,759 disclosed an apparatus using a hardware element for estimating jitter in the voice packets over a network. The network follows the TCP/IP protocol. [0010] U.S. Pat. No. 6,700,895 disclosed a method for determining the optimal jitter buffer size based on the data packet loss in a real-time communication system. [0011] U.S. Pat. No. 6,683,889 disclosed a method for automatically adjusting the jitter buffer size. The method determines the jitter buffer size by comparing the packet delay and a default value. [0012] However, the estimation of the network delay remains difficult. The conventional techniques use the time stamp on the voice packet to compute the network delay, which may also be affected by the clock rate discrepancy between the transmitting and receiving ends. Therefore, the sampling rate and the communication may not be synchronized. The sampling rate discrepancy may be a result of the hardware at the transmission and receiving end. For example, the voice sampling is configured to be 8 KHz. The software is based on 8 KHz to encode and decode the voice signals. However, if the hardware devices at both ends are not exactly setting at 8 KHz, the error will occur. [0013] The aforementioned techniques fail to effectively solve the problem of estimating the voice packet playout delay. Some techniques require extra hardware element for implementation, while others do not support silence adjustment to adjust the playout time. However, the voice packet playout delay is the key to the quality. SUMMARY OF THE INVENTION [0014] The present invention has been made to overcome the above-mentioned drawback of conventional methods. The primary object of the present invention is to provide a method and apparatus for dynamically adjusting the playout delay of audio signals to reduce the impact of the network delay variation on the voice quality and improve the voice smoothness. [0015] The method for dynamically adjusting the playout delay of audio signals of the present invention includes three dynamic adjustment parts: (a) dynamic adjustment of playout delay, (b) dynamic adjustment of the silence length, and (c) dynamic adjustment of jitter buffer zone. The best time for the (a) dynamic adjustment of playout delay is during the silence. The silence length in (b) is determined by the number of the voice packets in the jitter buffer. The zone size in (c) depends on the number of the voice packets in the jitter buffer. [0016] According to the present invention, the playout delay is adjusted in real time in accordance with the distribution of the number of the voice packets in the jitter buffer. A voice active detection (VAD) mechanism is used at the receiving end to detect the silence in the voice packets. By adjusting the silence length in the voice packets to change the playout delay, the impact of the network variation on the voice quality is reduced. [0017] The jitter buffer is divided into five different zones by three boundaries. The three boundaries are the lower bound of normal delay, the upper bound of normal delay and the maximum acceptable delay. The maximum acceptable delay is the maximum delay that is acceptable during the voice conversation. [0018] When the amount of the voice packets in jitter buffer exceeds the maximum acceptable delay, the jitter buffer discards the voice packets beyond the boundary. When the amount of the voice packets in jitter buffer is between the maximum acceptable delay and the upper bound of normal delay, it indicates the amount of voice packets in the jitter buffer is too large but still within the storage limit. The VAD is activated to detect the silence in the voice packets and shrink the silence length to reduce the playout delay. If the amount of the voice packets in jitter buffer is between upper bound of normal delay and the lower bound of normal delay, it indicates the amount of the voice packets in jitter buffer is within the acceptable range. No further processing is required. When the amount of the voice packets in jitter buffer is lower than the lower bound of normal delay, it indicates the amount of the voice packets in jitter buffer is too small but there remain voice packets for playout. The VAD is activated to detect the silence in the voice packets and extend the silence length to increase the playout delay. [0019] Other than the condition when the amount of voice packets in the jitter buffer is between the upper bound of normal delay and lower bound of normal delay, all the voice packets are processed before they are played out. The best scenario is that all the voice packets can be played out without processing, that is, without adjusting the silence length. To achieve the object, the present invention adjusts the zone size according to the distribution of the probabilities of the voice packet amount falls within the zones. Through a probability model to estimate the network variation and an algorithm for adjusting the zones, the zones can be automatically adjusted according to the network conditions. [0020] Therefore, the apparatus using the method of the present invention includes a jitter buffer, a dynamic playback delay adjustment module, a dynamic silence length adjustment module, and a dynamic jitter buffer zone adjustment module. The jitter buffer further includes an extended silence zone, a normal delay range zone, and a shrink silence zone. The dynamic jitter buffer zone adjustment module further includes a probability model estimation unit and a zone size adjustment module. [0021] The present invention reduces the probability for processing voice packets before playout so that the quality of the voice is better ensured and the amount of total computation is reduced. Continue reading about Method and apparatus for dynamically adjusting the playout delay of audio signals... 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